In the usual PCM telephone networks, voice samples are quantized in the form of 8-bit code words which are assembled in frames divided into a multiplicity of time slots, each time slot encompassing eight basic clock cycles or bit periods so as to accommodate an 8-bit word. Thus, in conformity with European telecommunication standards, a frame may consist of 32 time slots of which all but a few are allotted to message code words, with the remainder reserved for supervisory signal codes. Allowing for a synchronization code and for an identification code, such a frame may carry up to 30 8-bit code words representing as many PCM channels.
If two bit streams consisting of recurrent PCM frames arrive simultaneously at one terminal from which they are to be transmitted to another via a common PCM link, conventional techniques would require a doubling of the bit rate along that common path if the full information carried by the original code words is to be conveyed.
Recently there has been developed, especially for the transmission of voice samples in a telephone system, a technique known as adaptive differential pulse-code modulation (ADPCM) which utilizes predetermined characteristics of the human voice spectrum for short-term prediction of the magnitude of the next sample and quantizes the error, if any, between the actual and predicted samples. The quantization is exponentially weighted under the control of the quantized error signal itself. Reference in this connection may be made to commonly owned Italian patent No. 984,398 and to a paper (CIV-23) published by us in March 1974 in the Proceedings of the 21.sup.st International Electronic and Nuclear Congress of Rome, entitled PROTOTIPO DI LABORATORIO DI PCM DIFFERENZIALE CON ADATTAMENTO DEL PASSO DI QUANTIZZAZIONE PER TRANSMISSIONI VOCALI (Laboratory Prototype of Differential PCM with Adaptation of the Quantization Rate of Voice Transmission).